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-rw-r--r--sound/core/compress_offload.c3
-rw-r--r--sound/core/pcm_compat.c1
-rw-r--r--sound/core/seq/seq_clientmgr.c6
-rw-r--r--sound/core/seq/seq_lock.c4
-rw-r--r--sound/core/seq/seq_lock.h12
-rw-r--r--sound/core/seq/seq_ports.c7
-rw-r--r--sound/core/seq/seq_virmidi.c27
-rw-r--r--sound/core/vmaster.c31
-rw-r--r--sound/hda/hdac_controller.c5
-rw-r--r--sound/pci/asihpi/hpioctl.c12
-rw-r--r--sound/pci/echoaudio/echoaudio.c6
-rw-r--r--sound/pci/hda/hda_codec.c97
-rw-r--r--sound/pci/hda/patch_hdmi.c21
-rw-r--r--sound/pci/hda/patch_realtek.c19
-rw-r--r--sound/soc/codecs/adau17x1.c24
-rw-r--r--sound/soc/codecs/adau17x1.h2
-rw-r--r--sound/soc/codecs/rt5514-spi.c17
-rw-r--r--sound/soc/codecs/rt5514-spi.h3
-rw-r--r--sound/soc/codecs/rt5514.c63
-rw-r--r--sound/soc/codecs/rt5514.h3
-rw-r--r--sound/soc/codecs/rt5616.c2
-rw-r--r--sound/soc/codecs/rt5659.c4
-rw-r--r--sound/soc/soc-topology.c5
-rw-r--r--sound/usb/caiaq/device.c12
-rw-r--r--sound/usb/card.c20
-rw-r--r--sound/usb/line6/driver.c7
-rw-r--r--sound/usb/line6/podhd.c8
-rw-r--r--sound/usb/mixer.c12
-rw-r--r--sound/usb/mixer.h2
-rw-r--r--sound/usb/quirks.c4
-rw-r--r--sound/usb/usx2y/usb_stream.c6
31 files changed, 282 insertions, 163 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index fec1dfdb14ad..4490a699030b 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -948,14 +948,13 @@ static const struct file_operations snd_compr_file_ops = {
static int snd_compress_dev_register(struct snd_device *device)
{
int ret = -EINVAL;
- char str[16];
struct snd_compr *compr;
if (snd_BUG_ON(!device || !device->device_data))
return -EBADFD;
compr = device->device_data;
- pr_debug("reg %s for device %s, direction %d\n", str, compr->name,
+ pr_debug("reg device %s, direction %d\n", compr->name,
compr->direction);
/* register compressed device */
ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS,
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 3a1cc7b97e46..b719d0bd833e 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -547,6 +547,7 @@ struct snd_pcm_mmap_status_x32 {
u32 pad2; /* alignment */
struct timespec tstamp;
s32 suspended_state;
+ s32 pad3;
struct timespec audio_tstamp;
} __packed;
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index ea2d0ae85bd3..6c9cba2166d9 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1259,6 +1259,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg)
struct snd_seq_port_info *info = arg;
struct snd_seq_client_port *port;
struct snd_seq_port_callback *callback;
+ int port_idx;
/* it is not allowed to create the port for an another client */
if (info->addr.client != client->number)
@@ -1269,7 +1270,9 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg)
return -ENOMEM;
if (client->type == USER_CLIENT && info->kernel) {
- snd_seq_delete_port(client, port->addr.port);
+ port_idx = port->addr.port;
+ snd_seq_port_unlock(port);
+ snd_seq_delete_port(client, port_idx);
return -EINVAL;
}
if (client->type == KERNEL_CLIENT) {
@@ -1290,6 +1293,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg)
snd_seq_set_port_info(port, info);
snd_seq_system_client_ev_port_start(port->addr.client, port->addr.port);
+ snd_seq_port_unlock(port);
return 0;
}
diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c
index 0ff7926a5a69..cda64b489e42 100644
--- a/sound/core/seq/seq_lock.c
+++ b/sound/core/seq/seq_lock.c
@@ -23,8 +23,6 @@
#include <sound/core.h>
#include "seq_lock.h"
-#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG)
-
/* wait until all locks are released */
void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line)
{
@@ -41,5 +39,3 @@ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line)
}
}
EXPORT_SYMBOL(snd_use_lock_sync_helper);
-
-#endif
diff --git a/sound/core/seq/seq_lock.h b/sound/core/seq/seq_lock.h
index 54044bc2c9ef..ac38031c370e 100644
--- a/sound/core/seq/seq_lock.h
+++ b/sound/core/seq/seq_lock.h
@@ -3,8 +3,6 @@
#include <linux/sched.h>
-#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG)
-
typedef atomic_t snd_use_lock_t;
/* initialize lock */
@@ -20,14 +18,4 @@ typedef atomic_t snd_use_lock_t;
void snd_use_lock_sync_helper(snd_use_lock_t *lock, const char *file, int line);
#define snd_use_lock_sync(lockp) snd_use_lock_sync_helper(lockp, __BASE_FILE__, __LINE__)
-#else /* SMP || CONFIG_SND_DEBUG */
-
-typedef spinlock_t snd_use_lock_t; /* dummy */
-#define snd_use_lock_init(lockp) /**/
-#define snd_use_lock_use(lockp) /**/
-#define snd_use_lock_free(lockp) /**/
-#define snd_use_lock_sync(lockp) /**/
-
-#endif /* SMP || CONFIG_SND_DEBUG */
-
#endif /* __SND_SEQ_LOCK_H */
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 0a7020c82bfc..d21ece9f8d73 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -122,7 +122,9 @@ static void port_subs_info_init(struct snd_seq_port_subs_info *grp)
}
-/* create a port, port number is returned (-1 on failure) */
+/* create a port, port number is returned (-1 on failure);
+ * the caller needs to unref the port via snd_seq_port_unlock() appropriately
+ */
struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client,
int port)
{
@@ -151,6 +153,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client,
snd_use_lock_init(&new_port->use_lock);
port_subs_info_init(&new_port->c_src);
port_subs_info_init(&new_port->c_dest);
+ snd_use_lock_use(&new_port->use_lock);
num = port >= 0 ? port : 0;
mutex_lock(&client->ports_mutex);
@@ -165,9 +168,9 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client,
list_add_tail(&new_port->list, &p->list);
client->num_ports++;
new_port->addr.port = num; /* store the port number in the port */
+ sprintf(new_port->name, "port-%d", num);
write_unlock_irqrestore(&client->ports_lock, flags);
mutex_unlock(&client->ports_mutex);
- sprintf(new_port->name, "port-%d", num);
return new_port;
}
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 8d93a4021c78..f48a4cd24ffc 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -77,13 +77,17 @@ static void snd_virmidi_init_event(struct snd_virmidi *vmidi,
* decode input event and put to read buffer of each opened file
*/
static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
- struct snd_seq_event *ev)
+ struct snd_seq_event *ev,
+ bool atomic)
{
struct snd_virmidi *vmidi;
unsigned char msg[4];
int len;
- read_lock(&rdev->filelist_lock);
+ if (atomic)
+ read_lock(&rdev->filelist_lock);
+ else
+ down_read(&rdev->filelist_sem);
list_for_each_entry(vmidi, &rdev->filelist, list) {
if (!vmidi->trigger)
continue;
@@ -97,7 +101,10 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
snd_rawmidi_receive(vmidi->substream, msg, len);
}
}
- read_unlock(&rdev->filelist_lock);
+ if (atomic)
+ read_unlock(&rdev->filelist_lock);
+ else
+ up_read(&rdev->filelist_sem);
return 0;
}
@@ -115,7 +122,7 @@ int snd_virmidi_receive(struct snd_rawmidi *rmidi, struct snd_seq_event *ev)
struct snd_virmidi_dev *rdev;
rdev = rmidi->private_data;
- return snd_virmidi_dev_receive_event(rdev, ev);
+ return snd_virmidi_dev_receive_event(rdev, ev, true);
}
#endif /* 0 */
@@ -130,7 +137,7 @@ static int snd_virmidi_event_input(struct snd_seq_event *ev, int direct,
rdev = private_data;
if (!(rdev->flags & SNDRV_VIRMIDI_USE))
return 0; /* ignored */
- return snd_virmidi_dev_receive_event(rdev, ev);
+ return snd_virmidi_dev_receive_event(rdev, ev, atomic);
}
/*
@@ -209,7 +216,6 @@ static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream)
struct snd_virmidi_dev *rdev = substream->rmidi->private_data;
struct snd_rawmidi_runtime *runtime = substream->runtime;
struct snd_virmidi *vmidi;
- unsigned long flags;
vmidi = kzalloc(sizeof(*vmidi), GFP_KERNEL);
if (vmidi == NULL)
@@ -223,9 +229,11 @@ static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream)
vmidi->client = rdev->client;
vmidi->port = rdev->port;
runtime->private_data = vmidi;
- write_lock_irqsave(&rdev->filelist_lock, flags);
+ down_write(&rdev->filelist_sem);
+ write_lock_irq(&rdev->filelist_lock);
list_add_tail(&vmidi->list, &rdev->filelist);
- write_unlock_irqrestore(&rdev->filelist_lock, flags);
+ write_unlock_irq(&rdev->filelist_lock);
+ up_write(&rdev->filelist_sem);
vmidi->rdev = rdev;
return 0;
}
@@ -264,9 +272,11 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream)
struct snd_virmidi_dev *rdev = substream->rmidi->private_data;
struct snd_virmidi *vmidi = substream->runtime->private_data;
+ down_write(&rdev->filelist_sem);
write_lock_irq(&rdev->filelist_lock);
list_del(&vmidi->list);
write_unlock_irq(&rdev->filelist_lock);
+ up_write(&rdev->filelist_sem);
snd_midi_event_free(vmidi->parser);
substream->runtime->private_data = NULL;
kfree(vmidi);
@@ -520,6 +530,7 @@ int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmi
rdev->rmidi = rmidi;
rdev->device = device;
rdev->client = -1;
+ init_rwsem(&rdev->filelist_sem);
rwlock_init(&rdev->filelist_lock);
INIT_LIST_HEAD(&rdev->filelist);
rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH;
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 6c58e6f73a01..e43af18d4383 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -484,3 +484,34 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
master->hook(master->hook_private_data, master->val);
}
EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster);
+
+/**
+ * snd_ctl_apply_vmaster_slaves - Apply function to each vmaster slave
+ * @kctl: vmaster kctl element
+ * @func: function to apply
+ * @arg: optional function argument
+ *
+ * Apply the function @func to each slave kctl of the given vmaster kctl.
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
+ int (*func)(struct snd_kcontrol *, void *),
+ void *arg)
+{
+ struct link_master *master;
+ struct link_slave *slave;
+ int err;
+
+ master = snd_kcontrol_chip(kctl);
+ err = master_init(master);
+ if (err < 0)
+ return err;
+ list_for_each_entry(slave, &master->slaves, list) {
+ err = func(&slave->slave, arg);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_slaves);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 978dc1801b3a..f6d2985b2520 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -284,6 +284,11 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus)
dev_dbg(bus->dev, "HDA capability ID: 0x%x\n",
(cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF);
+ if (cur_cap == -1) {
+ dev_dbg(bus->dev, "Invalid capability reg read\n");
+ break;
+ }
+
switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) {
case AZX_ML_CAP_ID:
dev_dbg(bus->dev, "Found ML capability\n");
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 7e3aa50b21f9..5badd08e1d69 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -103,6 +103,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
void __user *puhr;
union hpi_message_buffer_v1 *hm;
union hpi_response_buffer_v1 *hr;
+ u16 msg_size;
u16 res_max_size;
u32 uncopied_bytes;
int err = 0;
@@ -127,22 +128,25 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
}
/* Now read the message size and data from user space. */
- if (get_user(hm->h.size, (u16 __user *)puhm)) {
+ if (get_user(msg_size, (u16 __user *)puhm)) {
err = -EFAULT;
goto out;
}
- if (hm->h.size > sizeof(*hm))
- hm->h.size = sizeof(*hm);
+ if (msg_size > sizeof(*hm))
+ msg_size = sizeof(*hm);
/* printk(KERN_INFO "message size %d\n", hm->h.wSize); */
- uncopied_bytes = copy_from_user(hm, puhm, hm->h.size);
+ uncopied_bytes = copy_from_user(hm, puhm, msg_size);
if (uncopied_bytes) {
HPI_DEBUG_LOG(ERROR, "uncopied bytes %d\n", uncopied_bytes);
err = -EFAULT;
goto out;
}
+ /* Override h.size in case it is changed between two userspace fetches */
+ hm->h.size = msg_size;
+
if (get_user(res_max_size, (u16 __user *)puhr)) {
err = -EFAULT;
goto out;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 7326695bca33..d68f99e076a8 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = ECHOGAIN_MAXOUT;
uinfo->dimen.d[0] = num_busses_out(chip);
uinfo->dimen.d[1] = num_busses_in(chip);
- uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1];
return 0;
}
@@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = ECHOGAIN_MAXOUT;
uinfo->dimen.d[0] = num_busses_out(chip);
uinfo->dimen.d[1] = num_pipes_out(chip);
- uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1];
return 0;
}
@@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 96;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = 0;
#ifdef ECHOCARD_HAS_VMIXER
@@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
#endif
uinfo->dimen.d[1] = 16; /* 16 channels */
uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */
- uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2];
return 0;
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3db26c451837..a0989d231fd0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1803,36 +1803,6 @@ static int check_slave_present(struct hda_codec *codec,
return 1;
}
-/* guess the value corresponding to 0dB */
-static int get_kctl_0dB_offset(struct hda_codec *codec,
- struct snd_kcontrol *kctl, int *step_to_check)
-{
- int _tlv[4];
- const int *tlv = NULL;
- int val = -1;
-
- if ((kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) &&
- kctl->tlv.c == snd_hda_mixer_amp_tlv) {
- get_ctl_amp_tlv(kctl, _tlv);
- tlv = _tlv;
- } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
- tlv = kctl->tlv.p;
- if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) {
- int step = tlv[3];
- step &= ~TLV_DB_SCALE_MUTE;
- if (!step)
- return -1;
- if (*step_to_check && *step_to_check != step) {
- codec_err(codec, "Mismatching dB step for vmaster slave (%d!=%d)\n",
-- *step_to_check, step);
- return -1;
- }
- *step_to_check = step;
- val = -tlv[2] / step;
- }
- return val;
-}
-
/* call kctl->put with the given value(s) */
static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
{
@@ -1847,19 +1817,58 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
return 0;
}
-/* initialize the slave volume with 0dB */
-static int init_slave_0dB(struct hda_codec *codec,
- void *data, struct snd_kcontrol *slave)
+struct slave_init_arg {
+ struct hda_codec *codec;
+ int step;
+};
+
+/* initialize the slave volume with 0dB via snd_ctl_apply_vmaster_slaves() */
+static int init_slave_0dB(struct snd_kcontrol *kctl, void *_arg)
{
- int offset = get_kctl_0dB_offset(codec, slave, data);
- if (offset > 0)
- put_kctl_with_value(slave, offset);
+ struct slave_init_arg *arg = _arg;
+ int _tlv[4];
+ const int *tlv = NULL;
+ int step;
+ int val;
+
+ if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
+ if (kctl->tlv.c != snd_hda_mixer_amp_tlv) {
+ codec_err(arg->codec,
+ "Unexpected TLV callback for slave %s:%d\n",
+ kctl->id.name, kctl->id.index);
+ return 0; /* ignore */
+ }
+ get_ctl_amp_tlv(kctl, _tlv);
+ tlv = _tlv;
+ } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
+ tlv = kctl->tlv.p;
+
+ if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_SCALE)
+ return 0;
+
+ step = tlv[3];
+ step &= ~TLV_DB_SCALE_MUTE;
+ if (!step)
+ return 0;
+ if (arg->step && arg->step != step) {
+ codec_err(arg->codec,
+ "Mismatching dB step for vmaster slave (%d!=%d)\n",
+ arg->step, step);
+ return 0;
+ }
+
+ arg->step = step;
+ val = -tlv[2] / step;
+ if (val > 0) {
+ put_kctl_with_value(kctl, val);
+ return val;
+ }
+
return 0;
}
-/* unmute the slave */
-static int init_slave_unmute(struct hda_codec *codec,
- void *data, struct snd_kcontrol *slave)
+/* unmute the slave via snd_ctl_apply_vmaster_slaves() */
+static int init_slave_unmute(struct snd_kcontrol *slave, void *_arg)
{
return put_kctl_with_value(slave, 1);
}
@@ -1919,9 +1928,13 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
/* init with master mute & zero volume */
put_kctl_with_value(kctl, 0);
if (init_slave_vol) {
- int step = 0;
- map_slaves(codec, slaves, suffix,
- tlv ? init_slave_0dB : init_slave_unmute, &step);
+ struct slave_init_arg arg = {
+ .codec = codec,
+ .step = 0,
+ };
+ snd_ctl_apply_vmaster_slaves(kctl,
+ tlv ? init_slave_0dB : init_slave_unmute,
+ &arg);
}
if (ctl_ret)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2b64fabd5faa..c19c81d230bd 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
hda_nid_t pin_nid, u32 stream_tag, int format)
{
struct hdmi_spec *spec = codec->spec;
+ unsigned int param;
int err;
err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format));
@@ -915,6 +916,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
return err;
}
+ if (is_haswell_plus(codec)) {
+
+ /*
+ * on recent platforms IEC Coding Type is required for HBR
+ * support, read current Digital Converter settings and set
+ * ICT bitfield if needed.
+ */
+ param = snd_hda_codec_read(codec, cvt_nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0);
+
+ param = (param >> 16) & ~(AC_DIG3_ICT);
+
+ /* on recent platforms ICT mode is required for HBR support */
+ if (is_hbr_format(format))
+ param |= 0x1;
+
+ snd_hda_codec_write(codec, cvt_nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_3, param);
+ }
+
snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format);
return 0;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ce71111b4e3..546d515f3c1f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -327,6 +327,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0215:
case 0x10ec0225:
case 0x10ec0233:
+ case 0x10ec0236:
case 0x10ec0255:
case 0x10ec0256:
case 0x10ec0282:
@@ -911,6 +912,7 @@ static struct alc_codec_rename_pci_table rename_pci_tbl[] = {
{ 0x10ec0275, 0x1028, 0, "ALC3260" },
{ 0x10ec0899, 0x1028, 0, "ALC3861" },
{ 0x10ec0298, 0x1028, 0, "ALC3266" },
+ { 0x10ec0236, 0x1028, 0, "ALC3204" },
{ 0x10ec0256, 0x1028, 0, "ALC3246" },
{ 0x10ec0225, 0x1028, 0, "ALC3253" },
{ 0x10ec0295, 0x1028, 0, "ALC3254" },
@@ -3930,6 +3932,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
alc_process_coef_fw(codec, coef0255_1);
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
alc_process_coef_fw(codec, coef0255);
@@ -4028,6 +4031,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
};
switch (codec->core.vendor_id) {
+ case 0x10ec0236:
case 0x10ec0255:
case 0x10ec0256:
alc_write_coef_idx(codec, 0x45, 0xc489);
@@ -4160,6 +4164,7 @@ static void alc_headset_mode_default(struct hda_codec *codec)
alc_process_coef_fw(codec, alc225_pre_hsmode);
alc_process_coef_fw(codec, coef0225);
break;
+ case 0x10ec0236:
case 0x10ec0255:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
@@ -4256,6 +4261,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
break;
@@ -4366,6 +4372,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
break;
@@ -4451,6 +4458,7 @@ static void alc_determine_headset_type(struct hda_codec *codec)
};
switch (codec->core.vendor_id) {
+ case 0x10ec0236:
case 0x10ec0255:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
@@ -4705,6 +4713,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec)
case 0x10ec0255:
alc_process_coef_fw(codec, alc255fw);
break;
+ case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, alc256fw);
break;
@@ -6419,6 +6428,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
ALC225_STANDARD_PINS,
{0x12, 0xb7a60130},
{0x1b, 0x90170110}),
+ SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x12, 0x90a60140},
+ {0x14, 0x90170110},
+ {0x21, 0x02211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x12, 0x90a60140},
+ {0x14, 0x90170150},
+ {0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
{0x14, 0x90170110},
{0x21, 0x02211020}),
@@ -6806,6 +6823,7 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0255:
spec->codec_variant = ALC269_TYPE_ALC255;
break;
+ case 0x10ec0236:
case 0x10ec0256:
spec->codec_variant = ALC269_TYPE_ALC256;
spec->shutup = alc256_shutup;
@@ -7857,6 +7875,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260),
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 2c1bd2763864..6758f789b712 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -90,6 +90,27 @@ static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int adau17x1_adc_fixup(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct adau *adau = snd_soc_codec_get_drvdata(codec);
+
+ /*
+ * If we are capturing, toggle the ADOSR bit in Converter Control 0 to
+ * avoid losing SNR (workaround from ADI). This must be done after
+ * the ADC(s) have been enabled. According to the data sheet, it is
+ * normally illegal to set this bit when the sampling rate is 96 kHz,
+ * but according to ADI it is acceptable for this workaround.
+ */
+ regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0,
+ ADAU17X1_CONVERTER0_ADOSR, ADAU17X1_CONVERTER0_ADOSR);
+ regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0,
+ ADAU17X1_CONVERTER0_ADOSR, 0);
+
+ return 0;
+}
+
static const char * const adau17x1_mono_stereo_text[] = {
"Stereo",
"Mono Left Channel (L+R)",
@@ -121,7 +142,8 @@ static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0,
&adau17x1_dac_mode_mux),
- SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0),
+ SND_SOC_DAPM_ADC_E("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0,
+ adau17x1_adc_fixup, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0),
SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0),
SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0),
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index bf04b7efee40..db350035fad7 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -129,5 +129,7 @@ bool adau17x1_has_dsp(struct adau *adau);
#define ADAU17X1_CONVERTER0_CONVSR_MASK 0x7
+#define ADAU17X1_CONVERTER0_ADOSR BIT(3)
+
#endif
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index ed6e5373916c..12f2ecf3a4fe 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -145,9 +145,8 @@ done:
mutex_unlock(&rt5514_dsp->dma_lock);
}
-static irqreturn_t rt5514_spi_irq(int irq, void *data)
+static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
{
- struct rt5514_dsp *rt5514_dsp = data;
u8 buf[8];
rt5514_dsp->get_size = 0;
@@ -180,6 +179,13 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data)
if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
schedule_delayed_work(&rt5514_dsp->copy_work, 0);
+}
+
+static irqreturn_t rt5514_spi_irq(int irq, void *data)
+{
+ struct rt5514_dsp *rt5514_dsp = data;
+
+ rt5514_schedule_copy(rt5514_dsp);
return IRQ_HANDLED;
}
@@ -199,12 +205,19 @@ static int rt5514_spi_hw_params(struct snd_pcm_substream *substream,
struct rt5514_dsp *rt5514_dsp =
snd_soc_platform_get_drvdata(rtd->platform);
int ret;
+ u8 buf[8];
mutex_lock(&rt5514_dsp->dma_lock);
ret = snd_pcm_lib_alloc_vmalloc_buffer(substream,
params_buffer_bytes(hw_params));
rt5514_dsp->substream = substream;
rt5514_dsp->dma_offset = 0;
+
+ /* Read IRQ status and schedule copy accordingly. */
+ rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf));
+ if (buf[0] & RT5514_IRQ_STATUS_BIT)
+ rt5514_schedule_copy(rt5514_dsp);
+
mutex_unlock(&rt5514_dsp->dma_lock);
return ret;
diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h
index a6434ee6ff03..c1a36647c119 100644
--- a/sound/soc/codecs/rt5514-spi.h
+++ b/sound/soc/codecs/rt5514-spi.h
@@ -20,6 +20,9 @@
#define RT5514_BUFFER_VOICE_BASE 0x18000200
#define RT5514_BUFFER_VOICE_LIMIT 0x18000204
#define RT5514_BUFFER_VOICE_WP 0x1800020c
+#define RT5514_IRQ_CTRL 0x18002094
+
+#define RT5514_IRQ_STATUS_BIT (0x1 << 5)
/* SPI Command */
enum {
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 0945d212b8dc..d7956ababd11 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -338,39 +338,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
fw = NULL;
}
- if (rt5514->model_buf && rt5514->model_len) {
-#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
- int ret;
-
- ret = rt5514_spi_burst_write(0x4ff80000,
- rt5514->model_buf,
- ((rt5514->model_len / 8) + 1) * 8);
- if (ret) {
- dev_err(codec->dev,
- "Model load failed %d\n", ret);
- return ret;
- }
-#else
- dev_err(codec->dev,
- "No SPI driver for loading firmware\n");
-#endif
- } else {
- request_firmware(&fw, RT5514_FIRMWARE3,
- codec->dev);
- if (fw) {
-#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
- rt5514_spi_burst_write(0x4ff80000,
- fw->data,
- ((fw->size/8)+1)*8);
-#else
- dev_err(codec->dev,
- "No SPI driver to load fw\n");
-#endif
- release_firmware(fw);
- fw = NULL;
- }
- }
-
/* DSP run */
regmap_write(rt5514->i2c_regmap, 0x18002f00,
0x00055148);
@@ -385,34 +352,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
return 0;
}
-static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol,
- const unsigned int __user *bytes, unsigned int size)
-{
- struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
- struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
- struct snd_soc_codec *codec = rt5514->codec;
- int ret = 0;
-
- if (rt5514->model_buf || rt5514->model_len < size) {
- if (rt5514->model_buf)
- devm_kfree(codec->dev, rt5514->model_buf);
- rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL);
- if (!rt5514->model_buf) {
- ret = -ENOMEM;
- goto done;
- }
- }
-
- /* Skips the TLV header. */
- bytes += 2;
-
- if (copy_from_user(rt5514->model_buf, bytes, size))
- ret = -EFAULT;
-done:
- rt5514->model_len = (ret ? 0 : size);
- return ret;
-}
-
static const struct snd_kcontrol_new rt5514_snd_controls[] = {
SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST,
RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv),
@@ -424,8 +363,6 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = {
adc_vol_tlv),
SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0,
rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put),
- SND_SOC_BYTES_TLV("Hotword Model", 0x8504,
- NULL, rt5514_hotword_model_put),
};
/* ADC Mixer*/
diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h
index 803311cb7e2a..2dc40e6d8b3f 100644
--- a/sound/soc/codecs/rt5514.h
+++ b/sound/soc/codecs/rt5514.h
@@ -255,7 +255,6 @@
#define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin"
#define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin"
-#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin"
/* System Clock Source */
enum {
@@ -282,8 +281,6 @@ struct rt5514_priv {
int pll_in;
int pll_out;
int dsp_enabled;
- u8 *model_buf;
- unsigned int model_len;
};
#endif /* __RT5514_H__ */
diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index c94e94fe8297..0e5f54a9bc7e 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -98,7 +98,7 @@ static const struct reg_default rt5616_reg[] = {
{ 0x8e, 0x0004 },
{ 0x8f, 0x1100 },
{ 0x90, 0x0000 },
- { 0x91, 0x0000 },
+ { 0x91, 0x0c00 },
{ 0x92, 0x0000 },
{ 0x93, 0x2000 },
{ 0x94, 0x0200 },
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 71216db15eab..fa66b11df8d4 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -2744,7 +2744,8 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5659_hp_event,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
- SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_S("LOUT Amp", 1, RT5659_PWR_ANLG_1, RT5659_PWR_LM_BIT,
+ 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Charge Pump", SND_SOC_NOPM, 0, 0,
rt5659_charge_pump_event, SND_SOC_DAPM_PRE_PMU |
@@ -3208,6 +3209,7 @@ static const struct snd_soc_dapm_route rt5659_dapm_routes[] = {
{ "LOUT R MIX", "OUTVOL R Switch", "OUTVOL R" },
{ "LOUT Amp", NULL, "LOUT L MIX" },
{ "LOUT Amp", NULL, "LOUT R MIX" },
+ { "LOUT Amp", NULL, "Charge Pump" },
{ "LOUT Amp", NULL, "SYS CLK DET" },
{ "LOUT L Playback", "Switch", "LOUT Amp" },
{ "LOUT R Playback", "Switch", "LOUT Amp" },
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index dd471d2c0266..01a50413c66f 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1301,7 +1301,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
/* validate kcontrol */
if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
SNDRV_CTL_ELEM_ID_NAME_MAXLEN)
- return NULL;
+ goto err;
se = kzalloc(sizeof(*se), GFP_KERNEL);
if (se == NULL)
@@ -1378,6 +1378,9 @@ err_se:
for (; i >= 0; i--) {
/* free values and texts */
se = (struct soc_enum *)kc[i].private_value;
+ if (!se)
+ continue;
+
kfree(se->dobj.control.dvalues);
for (j = 0; j < ec->items; j++)
kfree(se->dobj.control.dtexts[j]);
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 0fb6b1b79261..d8409d9ae55b 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -469,10 +469,12 @@ static int init_card(struct snd_usb_caiaqdev *cdev)
err = snd_usb_caiaq_send_command(cdev, EP1_CMD_GET_DEVICE_INFO, NULL, 0);
if (err)
- return err;
+ goto err_kill_urb;
- if (!wait_event_timeout(cdev->ep1_wait_queue, cdev->spec_received, HZ))
- return -ENODEV;
+ if (!wait_event_timeout(cdev->ep1_wait_queue, cdev->spec_received, HZ)) {
+ err = -ENODEV;
+ goto err_kill_urb;
+ }
usb_string(usb_dev, usb_dev->descriptor.iManufacturer,
cdev->vendor_name, CAIAQ_USB_STR_LEN);
@@ -507,6 +509,10 @@ static int init_card(struct snd_usb_caiaqdev *cdev)
setup_card(cdev);
return 0;
+
+ err_kill_urb:
+ usb_kill_urb(&cdev->ep1_in_urb);
+ return err;
}
static int snd_probe(struct usb_interface *intf,
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 3dc36d913550..23d1d23aefec 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -221,6 +221,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
struct usb_interface_descriptor *altsd;
void *control_header;
int i, protocol;
+ int rest_bytes;
/* find audiocontrol interface */
host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0];
@@ -235,6 +236,15 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
return -EINVAL;
}
+ rest_bytes = (void *)(host_iface->extra + host_iface->extralen) -
+ control_header;
+
+ /* just to be sure -- this shouldn't hit at all */
+ if (rest_bytes <= 0) {
+ dev_err(&dev->dev, "invalid control header\n");
+ return -EINVAL;
+ }
+
switch (protocol) {
default:
dev_warn(&dev->dev,
@@ -245,11 +255,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
case UAC_VERSION_1: {
struct uac1_ac_header_descriptor *h1 = control_header;
+ if (rest_bytes < sizeof(*h1)) {
+ dev_err(&dev->dev, "too short v1 buffer descriptor\n");
+ return -EINVAL;
+ }
+
if (!h1->bInCollection) {
dev_info(&dev->dev, "skipping empty audio interface (v1)\n");
return -EINVAL;
}
+ if (rest_bytes < h1->bLength) {
+ dev_err(&dev->dev, "invalid buffer length (v1)\n");
+ return -EINVAL;
+ }
+
if (h1->bLength < sizeof(*h1) + h1->bInCollection) {
dev_err(&dev->dev, "invalid UAC_HEADER (v1)\n");
return -EINVAL;
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 0ff5a7d2e19f..c8f723c3a033 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -779,9 +779,10 @@ int line6_probe(struct usb_interface *interface,
return 0;
error:
- if (line6->disconnect)
- line6->disconnect(line6);
- snd_card_free(card);
+ /* we can call disconnect callback here because no close-sync is
+ * needed yet at this point
+ */
+ line6_disconnect(interface);
return ret;
}
EXPORT_SYMBOL_GPL(line6_probe);
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index 956f847a96e4..451007c27743 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -301,7 +301,8 @@ static void podhd_disconnect(struct usb_line6 *line6)
intf = usb_ifnum_to_if(line6->usbdev,
pod->line6.properties->ctrl_if);
- usb_driver_release_interface(&podhd_driver, intf);
+ if (intf)
+ usb_driver_release_interface(&podhd_driver, intf);
}
}
@@ -317,6 +318,9 @@ static int podhd_init(struct usb_line6 *line6,
line6->disconnect = podhd_disconnect;
+ init_timer(&pod->startup_timer);
+ INIT_WORK(&pod->startup_work, podhd_startup_workqueue);
+
if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) {
/* claim the data interface */
intf = usb_ifnum_to_if(line6->usbdev,
@@ -358,8 +362,6 @@ static int podhd_init(struct usb_line6 *line6,
}
/* init device and delay registering */
- init_timer(&pod->startup_timer);
- INIT_WORK(&pod->startup_work, podhd_startup_workqueue);
podhd_startup(pod);
return 0;
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 9732edf77f86..91bc8f18791e 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2234,6 +2234,9 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
static void snd_usb_mixer_free(struct usb_mixer_interface *mixer)
{
+ /* kill pending URBs */
+ snd_usb_mixer_disconnect(mixer);
+
kfree(mixer->id_elems);
if (mixer->urb) {
kfree(mixer->urb->transfer_buffer);
@@ -2584,8 +2587,13 @@ _error:
void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer)
{
- usb_kill_urb(mixer->urb);
- usb_kill_urb(mixer->rc_urb);
+ if (mixer->disconnected)
+ return;
+ if (mixer->urb)
+ usb_kill_urb(mixer->urb);
+ if (mixer->rc_urb)
+ usb_kill_urb(mixer->rc_urb);
+ mixer->disconnected = true;
}
#ifdef CONFIG_PM
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 2b4b067646ab..545d99b09706 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -22,6 +22,8 @@ struct usb_mixer_interface {
struct urb *rc_urb;
struct usb_ctrlrequest *rc_setup_packet;
u8 rc_buffer[6];
+
+ bool disconnected;
};
#define MAX_CHANNELS 16 /* max logical channels */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 913552078285..4f5f18f22974 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1137,6 +1137,9 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
case USB_ID(0x047F, 0x02F7): /* Plantronics BT-600 */
case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */
case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */
+ case USB_ID(0x047F, 0xC022): /* Plantronics C310 */
+ case USB_ID(0x047F, 0xC02F): /* Plantronics P610 */
+ case USB_ID(0x047F, 0xC036): /* Plantronics C520-M */
case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */
case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */
@@ -1351,6 +1354,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */
case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */
case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */
+ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */
if (fp->altsetting == 2)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 4dab49080700..e229abd21652 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -191,7 +191,8 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
}
pg = get_order(read_size);
- sk->s = (void *) __get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO, pg);
+ sk->s = (void *) __get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO|
+ __GFP_NOWARN, pg);
if (!sk->s) {
snd_printk(KERN_WARNING "couldn't __get_free_pages()\n");
goto out;
@@ -211,7 +212,8 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
pg = get_order(write_size);
sk->write_page =
- (void *)__get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO, pg);
+ (void *)__get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO|
+ __GFP_NOWARN, pg);
if (!sk->write_page) {
snd_printk(KERN_WARNING "couldn't __get_free_pages()\n");
usb_stream_free(sk);